sr-users March 2004

sr-users@lists.kamailio.org

145 participants
286 discussions

by Atle Samuelsen

Hey , I need to pull ut some stuff from the database, that's not default. Is it possible to use some function, that parces a sql query, and returns either True or false (1/0)? or do I have to write a small script, and use some exec function in ser to do this? - Atle

20 years, 1 month

RE: [Serusers] outgoing calls with alias

by Ralph.Wabel＠swisscom.com

My only intention is to associate a phone number to a user so that the called PSTN will see the correct phone number instead of the main phone number of the number block. Isn't there an easy way to make a translation of the username to a phone number? Or is for what I want to do the only way to make a radius server? Because it seems that for rpid you need to make a Sip-Rpid entry in /usr/local/etc/raddb/users Thanks a lot Ralph p.s. @jiri sorry I thought my mail goes to the usergroup and not only to you :-) -----Original Message----- From: Jiri Kuthan [mailto:jiri@iptel.org] Sent: Dienstag, 23. März 2004 10:50 To: Wabel Ralph, INO-ITA Subject: RE: [Serusers] outgoing calls with alias At 10:07 AM 3/23/2004, Ralph.Wabel(a)swisscom.com wrote: >Hi, >Thanks for the information. Do I really need a radius server or does it work without one? that depends on what you wish to accomplish. For authentication, you can use several database systems. >Is there a good tutorial about that? there is administrator's guide at SER webpage. -jiri >Thanks >Ralph > >-----Original Message----- >From: Jiri Kuthan [mailto:jiri@iptel.org] >Sent: Dienstag, 23. März 2004 00:46 >To: Wabel Ralph, INO-ITA; serusers(a)lists.iptel.org >Subject: Re: [Serusers] outgoing calls with alias > >Aliases have impact only on incoming calls, they don't mangle >outgoing signaling. They just rewrite r-uri, that's all they do. >To set callerid down in pstn, use rpid. > >-jiri > >At 05:42 PM 3/22/2004, Ralph.Wabel(a)swisscom.com wrote: >>Content-class: urn:content-classes:message >>Content-Type: multipart/alternative; >> boundary="----_=_NextPart_001_01C4102C.A43065FC" >> >>Hi, >> >>I have the problem that when I make an outgoing call over a Cisco Gateway to a PSTN phone. I've defined an alias for the user, works fine for incoming calls, but for the outgoing calls it shows always the number of the main number of the number block. When I make a debug I see that it goes out with the username instead of the alias. Here is my ser.cfg, maybe I've done something wrong in the config file. If someone could help me it would be great. Let me know if the config from the Cisco Gateway is also important. -- Jiri Kuthan http://iptel.org/~jiri/

20 years, 1 month

[Serusers] SER + B2BUA

by Kapil Dhawan

Hi all i don't want to use B2BUA but need the features of it....so want to write some small module for it.....can anyone suggest me how to proceed with it..... what all changes shud i look forward too and where to get the start for it regards _________________________________________________________________ Get head-hunted by 10,000 recruiters. http://go.msnserver.com/IN/44798.asp Post your CV on naukri.com today.

20 years, 1 month

[Serusers] ISDNgw + CAPI

by F.S.Salloum

Dear all, Has anybody installed ser with sems and isdngw? I checked the website and i found out a conf file (berlios) but has problems when it calls the vm module. The CAPI works fine with Asterisk PBX, but i prefer ser in order to develop more advanced hybrid services (HOLD, TRANSFER etc). Attached you can find the conf files. Thanks in advance Sotiris # $Id: isdngw.conf.sample,v 1.5 2003/11/16 19:24:56 ullstar Exp $ # # # Copyright (C) 2003 Ulrich Abend # # This file is part of isdngw, a free isdn gateway plugin # for sems, a free SIP media server. # # isdngw is free software; you can redistribute it and/or modify # it under the terms of the GNU General Public License as published by # the Free Software Foundation; either version 2 of the License, or # (at your option) any later version # # For a license to use the ser software under conditions # other than those described here, or to purchase support for this # software, please contact iptel.org by e-mail at the following addresses: # info(a)iptel.org # # isdngw is distributed in the hope that it will be useful, # but WITHOUT ANY WARRANTY; without even the implied warranty of # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the # GNU General Public License for more details. # # You should have received a copy of the GNU General Public License # along with this program; if not, write to the Free Software # Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA # # # isdngw.conf.sample # # ISDN Gateway Application Module for Sip Express Media Server (sems) # # sample configuration file # # # whitespaces (spaces and tabs) are ignored # comments start with a "#" and may be used inline # # example: option=value1, value2 # i like this option # # this file must be referenced in the global sems configuration file: # # sems.conf: # # isdngw module configuration (external file) # config.isdngw=/etc/isdngw.conf # ############################################################################### # backend selection: # # # # isdngw supports the classical isdn4linux standard by accessing the # # /dev/ttyI* devices as well as CAPI 2.0 based drivers. You can mix both # # technologies, e.g. using your classic fritzCard PCI with isdn4linux drivers # # and your C4 card with CAPI. # # # ############################################################################### # required parameter: enablei4l=< yes | no > # # enable I4L support? # For isdn4linux support, you need # - the linux kernel drivers for i4l loaded # - read/write permissions for sems user on (some) /dev/ttyI* devices # - you can (for whatever reason) use the "capidrv" kernel module # to use CAPI based cards with I4L (this is not recommended though) enablei4l=no # required parameter: enablecapi=< yes | no > # # enable CAPI20 support? # For CAPI support you need # - a sems compiled with CAPI support (-DUSE_CAPI20 uncommented in Makefile) # - the capi20 libraries from the isdn4k package properly installed # - the kernel capi drivers loaded (run capiinfo to check!) # CAPI20 support is experimental, tested with AVM FritzCard PCIV2 and C4 enablecapi=yes ############################################################################### # i4l backend configuration: # # # # If you enabled isdn4linux backend support above, you can configure the # # according parameters here. # # Backend selection for in- and outgoing calls is done according to your # # configuration. If a call matches the configuration of multiple backends # # the device selection is non-deterministic. # # # ############################################################################### # optional parameter: i4l_numoutdevices=<number of devices to be used> # # - specifies how many ttyI* devices to use for outgoing telephony calls # - devices must be fully accessible by the vm process' user # - this number also specified the maximum number of simultaneous # outgoing phone calls (if not otherwise restricted) # - you must specify exactly one parameter of i4l_numoutdevices and i4l_outdevices, not both i4l_numoutdevices=5 # optional parameter: i4l_outdevices=<dev1>, <dev2>, <dev3>, ... # # - specifies which ttyI* devices to use for outgoing telephony calls # - devices must be fully accessible by the vm process' user # - the number of devices listed is the maximum of simultaneous # outgoing phone calls (if not otherwise restricted) # - you may not set i4l_numoutdevices if this parameter is set # - e.g. i4l_outdevices=/dev/ttyI10, /dev/ttyI11, /dev/ttyI12 i4l_outdevices= # optional parameter: i4l_numindevices=<number of devices used> # # - specifies how many ttyI* devices to use for incoming telephony calls # - devices must be fully accessible by the sems process' user # - the number of devices listed is also the maximum of simultaneous # incoming phone calls (if not otherwise restricted) # - you may specify only i4l_numindevices OR i4l_indevices, not both. i4l_numindevices=5 # optional parameter: i4l_indevices=<dev1>, <dev2>, <dev3>, ... # # - specifies which ttyI* devices to use for incoming telephony calls # - devices must be fully accessible by the vm process' user # - the number of devices listed is the maximum of simultaneous # incoming phone calls (if not otherwise restricted) # - you may specify only i4l_numindevices OR i4l_indevices, not both. # - e.g. i4l_indevices=/dev/ttyI13, /dev/ttyI14 i4l_indevices= # optional parameter: i4l_lockdir=/where/to/store/lockfiles # # - specifies the directory where to put the lockfiles # - may be useful if not run as root # - default: i4l_lockdir=/var/lock i4l_lockdir=/var/lock # optional parameter: i4l_inmsn=<msn1>, <msn2>, <msn3>, ... # # - specifies the all msn's the i4l backend listens for calls on # - a call is only accepted if the according SIP call is successfully placed # - other ISDN equipment *may* also listen on this numbers, the # first to accept the call is the lucky one then # - IMPORTANT: no other isdn4linux application may bind to any of that numbers # - wildcards may be specified here like 81462* or * # - example: i4l_inmsn=12345, 54321* i4l_inmsn=* # optional parameter: i4l_mapping=<number1>, <number2>, ... # # - specifies what numbers are called via the isdn4linux backend # - a call is only placed to the PSTN network if it matches one mapping (i4l # or capi controller mapping below) # - wildcards may be specified here like 81462* or * # - A "*" means that all calls are allowed and get routed via i4l i4l_mapping=* # optional parameter: i4l_numchannel=<number of available b-channels> # # - specifies how many B-channels should be used with isdn4linux backend # - isdngw checks for availability of a channel before trying to route # the call, if another backend allows a connection and i4l has no # more channels available, other backend will be used # - the value does not have to be the real number of channels, you can # limit the max number of simulatenous connections with this parameter # - if this parameter is omitted, isdngw tries to detect this number # by accessing /dev/isdninfo, this fails if capidrv is used or the # user running sems does not have read permissions on this device i4l_numchannel= ############################################################################### # CAPI 2.0 backend configuration: # # # # If you enabled capi backend support above, you can configure the # # according parameters here. # # Backend selection for in- and outgoing calls is done according to your # # configuration. If a call matches the configuration of multiple backends # # the device selection is non-deterministic. # # # ############################################################################### # optional parameter: capi_inmsn=<msn1>, <msn2>, <msn3>, ... # # - specifies the all msn's the capi backend listens for calls on, this # setting is per controller based # - cards with multiple interfaces supply multiple # (virtual) controllers (e.g. AVM c4 supplies controller 1 to 4) # - you can use the capiinfo utility to see what controllers are available # - a call is only accepted if the according SIP call is successfully placed # - other ISDN equipment *may* also listen on this numbers, the # first to accept the call is the lucky one then # - wildcards may be specified here like 81462* or * # - isdngw currently supports up to 8 controllers # - example: capi_inmsn=12345, 54321* capi_1_inmsn=* capi_2_inmsn=* capi_3_inmsn=* capi_4_inmsn=* capi_5_inmsn=* capi_6_inmsn=* capi_7_inmsn=* capi_8_inmsn=* # optional parameter: capi_<ctrlnumber>_mapping=<number1>, <number2>, ... # # - specifies what numbers are called via each capi based controller backend # - this parameter must be specified for every controller to be used # (first controller: "capi_1_mapping" second controller: "capi_2_mapping" # - a call is only placed to the PSTN network if it matches one mapping (i4l # or capi controller mapping below) # - wildcards may be specified here like 81462* or * # - A "*" means that all calls are allowed and get routed via the specified # capi controller # - if multiple expressions match the longest wins, e.g. if controller 1 # handles 12345* and controller 2 handles 1*, controller 1 gets the calla # to 1234567 # - if you use the same expression for two or more controllers, the lower # controller number is tried first, if no line is available the other(s) # is tried. In case of other errors, the call is not tried more than once. # - isdngw currently supports up to 8 controllers # - example: # capi_1_mapping=0* # capi_2_mapping=8*, 9* # capi_3_mapping=1*, 2*, 3*, 4*, 5*, 6*, 7* capi_1_mapping=* capi_2_mapping=* capi_3_mapping=* capi_4_mapping=* capi_5_mapping=* capi_6_mapping=* capi_7_mapping=* capi_8_mapping=* # optional parameter: capi_<ctrlnumber>_numchannel=<number of available channels> # # - specifies how many B-channels should be used with each CAPI controller # - the value does not have to be the real number of channels, you can # limit the max number of simulatenous connections with this parameter # - if this parameter is omitted, isdngw detects the number of available # channels automatically # - example: # capi_1_numchannel=1 # capi_2_numchannel=1 # capi_3_numchannel= capi_1_numchannel= capi_2_numchannel= capi_3_numchannel= capi_4_numchannel= capi_5_numchannel= capi_6_numchannel= capi_7_numchannel= capi_8_numchannel= ############################################################################### # SIP configuration: # # # # Configure all SIP related settings here. # # # ############################################################################### # optional parameter: msnsipcaller=< yes | no > # # - specifies if the SIP caller username should be used as # outgoing msn (works only if SIP caller userpart is e164 style number) # - defaults to yes msnsipcaller=no # optional parameter: outmsn=<msn> # # - specifies the default msn for outgoing calls # - if msnsipcallee is set to "yes" this number is only used, # if SIP caller userpart is not e164 compliant # - if 0 is used, the number used depends on the configuration # of the PSTN telephony system defaultmsn=0 # required parameter: callerdomain=<domainname> # # - specifies the domainname that is used as initiator of the SIP calls # - the caller address is composed as follows: # <calling-number>@<callerdomain> #callerdomain=mycaller.domain.com #SSAL_ADITION callerdomain=146.124.2.235 # required parameter: calleedomain=<domainname> # # - specifies the domainname that is used as destination of the SIP calls # - the callee address is composed as follows: # <called-number>@<calleedomain> #calleedomain=mycallee.domain.com calleedomain=146.124.2.235 ############################################################################### # misc configuration: # # # # Other configuration options follow. # # # ############################################################################### # optional parameter: clip=< yes | no > # # - specifies whether Caller LIne Representation should be used # - no means hide caller number, yes means show caller number # - this works for CAPI but not for i4l, behaviour of i4l depends on # the configuration of the telephony system # - defaults to yes clip= # optional parameter: minnumberlen=<minimum lenght of number> # # - specifies the minimum number of digits on an outgoing call # - shorter numbers are rejected # - defaults to 4 minnumberlen= # optional parameter: outlogfile=<file> # # - specifies a log file, where all calls are listed # - WARNING: this function is unimplemented yet! outlogfile= # end of file. # $Id: sems.conf.sample,v 1.1.2.1 2003/08/28 19:13:25 rco Exp $ # # sems.conf.sample # # Sip Express Media Server (sems) # # sample configuration file # # # whitespaces (spaces and tabs) are ignored # comments start with a "#" and may be used inline # # example: option=value1, value2 # i like this option # ################################## # global parameters # ################################## # optional parameter: fork={yes|no} # # - specifies if sems should run in daemon mode (background) fork=yes # optional parameter: stderr={yes|no} # # - debug mode: do not fork and log to stderr stderr=no # optional parameter: loglevel={0|1|2|3} # # - sets log level (error=0, warning=1, info=2, debug=3) loglevel=1 # optional parameter: fifo_name=<filename> # # - path and file name of our fifo file fifo_name=/tmp/am_fifo # optional parameter: ser_fifo_name=<filename> # # - path and file name of Ser's fifo file ser_fifo_name=/tmp/ser_fifo # optional parameter: plugin_path=<path> # # - sets the path to the plug-ins # - may be absolute or relative to CWD plugin_path=/usr/local/lib/sems/plug-in/ ################################## # voicemail specific parameters # ################################## # optional parameter: announce_path=<path> # # - sets the path where announce files are searched for announce_path=/usr/local/lib/sems/audio/ # optional parameter: default_announce=<filename> # # - sets the name of the default announce WAV file default_announce=default_en.wav # optional parameter: max_record=<seconds> # # - maximum record time max_record=30 # optional parameter: smtp_server=<hostname> # # - sets address of smtp server smtp_server=localhost # optional parameter: smtp_port=<port> # # - sets port of smtp server smtp_port=25 ################################## # module specific parameters # ################################## # add more module configurations here (inline or external): # # config.mymodule=<filename> # or # config.mymodule=inline # ... # config.mymodule=end #isdngw module configuration (external file) config.isdngw=/root/ser/isdngw.conf # # $Id: ser-isdngw.conf,v 1.2 2003/09/09 17:48:22 ullstar Exp $ # # isdngw sample config script # please direct comments to ullstar(a)iptel.org # # ----------- global configuration parameters ------------------------ # setup parameters according to your needs. Most people will only have # to adjust the listen and alias parameters below. debug=1 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) check_via=yes # (cmd. line: -v) dns=0 # (cmd. line: -r) rev_dns=0 # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # Add the name of your system here #listen=yourhostname # for more names add alias entries, all that might be used as domain in SIP URI #alias=yourhostname.yourdomain.com #alias=your.ip.addr.ess # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/vm.so" loadmodule "/usr/local/lib/ser/modules/dbtext.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" # ----------------- setting module-specific parameters --------------- # You may want to define things like databases here. Please refer to # the extensive SER documentation for this purpose. Module parameters # are always described in the modules README files. # # For pure isdn gateway functionality only a database is needed, we use # a simple textfile for this purpose. Actually this is only neccessary until # the vm module is reworked. Simply copy the etc/db directory from the isdngw # directory somewhere and specify it in the following statement: modparam("voicemail", "db_url","/root/ser/db") # ------------------------- request routing logic ------------------- # This section describes how SIP messages are handled. # workaround needed for some buggy UAs (e.g. MS Messenger) modparam("rr", "enable_full_lr", 1) # The routing is described now: route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing loose_route(); # deal with my domain first if (uri==myself) { # the following allows for user-agent registration # in concerns of the isdn gateway this means: # register a SIP phone with <telephonenumber>@<this server name> # and configure the isdn gateway to listen on the number specified as # the user (<telephonenumber>). # Incoming calls are then directed to this user agent # Note: This sample configuration does not use a persistent storage, so # if you restart SER you have to re-register your SIP client to make this # work. Refer to the userloc module's documentation for setting up # persistent contact storage. if (method=="REGISTER") { save("location"); break; }; if (lookup("location")) { if (!t_relay()) { sl_reply_error(); }; break; }; # ############################## # # isdngw specific configuration # # ############################## # if(t_newtran()){ if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ # send a response right at the start to avoid retransmissions t_reply("100","Trying -- just wait a minute !"); # isdngw only gets activated on invite requests if(method=="INVITE"){ # isdngw feels to be responsible for numeric userparts # all numbers followed by @ and anything after it match # this expression # for example: sip:555123123@yourdomain.com:5061 matches. # The vm command (from module vm) is used to contact the # media server and though it the isdngw. # /tmp/am_fifo is the fifo filename ued for communications, make # shure the permissions are correct and that the same fifo # filename is defined in sems.conf. if(uri=~"sip:[0-9]+@.*"){ if(!vm("/tmp/am_fifo","isdngw")){ log("could not contact isdngw\n"); t_reply("500","could not contact isdngw"); }; # we dont feel responsible for sip addresses not starting with # a number, so send the right error code. } else { t_reply("404","Not Found"); }; # stop routing here, the message is now processed by the media server break; }; # The following handles the call termination, we must pass these requests # to the media server as follows. Again make shure the fifo name and permissions # are set correctly (like im sems.conf). if((method=="BYE")||(method=="CANCEL")){ if(!vm("/tmp/am_fifo","bye")){ log("could not contact the media server\n"); t_reply("500","could not contact the media server"); }; break; }; # other methods than INVITE, BYE and CANCEL are not handled by this SIP Server # so we sent an error message } else { log("ERROR: method not supported\n"); t_reply("500", "sorry, method not supported"); }; } else { # for any reason the transaction could not be created, send error code log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); }; # not uri=myself, this SIP request is not directed to us, simply direct it to its # correct destination } else { if (!t_relay()) { sl_reply_error(); }; }; # end of routing. }

20 years, 1 month

[Serusers] Radius client - ldconfig not adding the libarry

by Morris, Scott

I installed the radiusclient 0.4.1, and ran into a slight problem. I run ldconfig -v /usr/local/lib and the following is output: /usr/local/lib: libradiusclient.so.2 -> libradiusclient.so.2.0.0 Here is the directory of /usr/local/lib -rw-r--r-- 1 root root 130780 Mar 18 12:37 libradiusclient.a -rwxr-xr-x 1 root root 853 Mar 18 12:37 libradiusclient.la lrwxrwxrwx 1 root root 24 Mar 18 12:37 libradiusclient.so -> libradiusclient.so.2.0.0 lrwxrwxrwx 1 root root 24 Mar 18 12:37 libradiusclient.so.2 -> libradiusclient.so.2.0.0 -rwxr-xr-x 1 root root 95088 Mar 18 12:37 libradiusclient.so.2.0.0 Ldconfig will not process the libradiusclient.so because it is a symbolic link. So when I run "ser -ddddddddd start" I get the following: 0(7710) ERROR: load_module: could not open module </usr/lib/ser/modules/auth_radius.so>: libradiusclient.so.0: cannot open shared object file: No such file or directory Ser can't locate the reference for libradiusclient.so.0 since ldconfig won't process the link libradius.so.0. Any ideas of getting around this. I checked the man pages for ldconfig, and it states it does not process symbolic links: "ldconfig ignores symbolic links when scanning for libraries." Scott Morris Enterprise Network Engineer DOE - ORAU / ORISE 865-576-4672

20 years, 1 month

RE: [Serusers] radiusclient new generation, first beta release

by Morris, Scott

Is there a new HowTo on the Radisu installation? The one I found on iptel.org still address the radiusclient on www.mcs.de, and it doesn't exist any longer. I have also found the HowTo does not include all installationa dn configuration changes. Upon installing this I would be more than happy to add my installation notes. In trying to do this before, lfconfig was not documented in the install. I would add step by step if needed. Thanks. Scott Morris Enterprise Network Engineer DOE - ORAU / ORISE 865-576-4672 -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Tuesday, December 09, 2003 10:12 AM To: Maxim Sobolev Cc: jiri(a)iptel.org; serusers(a)lists.iptel.org; Daniel-Constantin.Mierla(a)fokus.fraunhofer.de Subject: Re: [Serusers] radiusclient new generation, first beta release Maxim, I like the changes, feel free to commit auth_radius, group_radius, and uri_radius patches. Regarding acc module, Jiri is the maintainer so he should say yes/no. I would like to ask you for one more thing, please update the ser-radius howto as well once you commit. Just a short note at the beginning that your version of radius library is required and where peope can get it should be enough. thanks, Jan.

20 years, 1 month

[Serusers] ser AAA with freeradius and mysql backend example

by Roddy G. Posada Santos

Hello everubody, Now i have compiled acc module with radius support, i have the accounting start but there is no accountig stop, exist any function to tell freeradius to stop accounting.

20 years, 1 month

[Serusers] serweb

by gaillac harry

Hello I have successfully installed and configured Serweb but cannot login to the web site. I enter admin / heslo and the web page just refreshes. I am not logged in. Can some one point me in the right direction? I am completely stumped. "register_globals=On" and "short_open_tag=On" in /etc/php,ini My configuration: Redhat Taroon updated ser-0.8.12-0 sems-0.1.1-0 php-4.3.2-7.ent mysql-3.23.56-1.9 mysql-server3.23.56-1.9 php-mysql-4.3.2-7 httpd-2.0.46-26.ent serweb_2004-01-04.tar.gz Best regards Harry

20 years, 1 month

[Serusers] ser AAA with freeradius and mysql backend example

by Roddy G. Posada Santos

Is this possible??? Actually i have ser (two instances) in the same linux box functioning ok with voicemail, now i want to try AAA with freeradius can anybody give some examples... i have freeradius with authorize/accounting in mysql database do you beleive that i can use the same with ser? Thanks Roddy

20 years, 1 month

RE: [Serusers] error loading acc module

by Roman Mikus

sorry I don't understand what are you pointing me to... >From that document I figured out, that function t_relay_to not longer exist in tm module. But I have problems with function t_relay_to_tcp. That function should be in tm. Is it correct? Why than I'm getting error message "tm_bind: TM module function 't_relay_to_tcp' not found"?? When I comment out line loadmodule "/usr/local/lib/ser/modules/acc.so" in my config, ser starts without complaining. I think that acc module is using t_relay_to_tcp, but from some reason it isn't found in tm. Thanks. Roman Mikus > -----Original Message----- > From: Jiri Kuthan [mailto:jiri@iptel.org] > Sent: Wednesday, March 17, 2004 2:22 PM > To: Roman Mikus; serusers(a)lists.iptel.org > Subject: Re: [Serusers] error loading acc module > > see http://lists.iptel.org/pipermail/serusers/2004-March/006655.html > > -jiri > > At 10:02 AM 3/17/2004, Roman Mikus wrote: > > >Hi, > > > >I'm running ser-0.8.12 on FreeBSD 4.9-RELEASE-p1. It was installed from > ports collection. Simple configuration works fine, until I try to load acc > module. Then I get following error message and ser server doesn't start. > > > > > >Mar 16 12:54:33 sirena /usr/local/sbin/ser[75391]: ERROR: tm_bind: TM > module function 't_relay_to_tcp' not found > >Mar 16 12:54:33 sirena /usr/local/sbin/ser[75391]: init_mod(): Error > while initializing module acc > > > >My ser config file is atached on the end. > > > >Can anybody help me with this? Thanks a lot. > > > >Roman Mikus > >mail to: roman(a)zutom.sk > > > ># > ># $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ > ># > ># simple quick-start config script > ># > > > ># ----------- global configuration parameters ------------------------ > > > >#debug=3 # debug level (cmd line: -dddddddddd) > >#fork=yes > >#log_stderror=no # (cmd line: -E) > > > >/* Uncomment these lines to enter debugging mode > >debug=7 > >fork=no > >log_stderror=yes > >*/ > > > >check_via=no # (cmd. line: -v) > >dns=no # (cmd. line: -r) > >rev_dns=no # (cmd. line: -R) > >#port=5060 > >#children=4 > >fifo="/tmp/ser_fifo" > > > ># ------------------ module loading ---------------------------------- > > > ># Uncomment this if you want to use SQL database > >loadmodule "/usr/local/lib/ser/modules/mysql.so" > > > >loadmodule "/usr/local/lib/ser/modules/tm.so" > >loadmodule "/usr/local/lib/ser/modules/acc.so" > >loadmodule "/usr/local/lib/ser/modules/sl.so" > >loadmodule "/usr/local/lib/ser/modules/rr.so" > >loadmodule "/usr/local/lib/ser/modules/maxfwd.so" > >loadmodule "/usr/local/lib/ser/modules/usrloc.so" > >loadmodule "/usr/local/lib/ser/modules/registrar.so" > > > ># Uncomment this if you want digest authentication > ># mysql.so must be loaded ! > >loadmodule "/usr/local/lib/ser/modules/auth.so" > >loadmodule "/usr/local/lib/ser/modules/auth_db.so" > > > ># ----------------- setting module-specific parameters --------------- > > > ># -- usrloc params -- > > > >#modparam("usrloc", "db_mode", 0) > > > ># Uncomment this if you want to use SQL database > ># for persistent storage and comment the previous line > >modparam("usrloc", "db_mode", 2) > > > ># -- auth params -- > ># Uncomment if you are using auth module > ># > >modparam("auth_db", "calculate_ha1", yes) > ># > ># If you set "calculate_ha1" parameter to yes (which true in this > config), > ># uncomment also the following parameter) > ># > >modparam("auth_db", "password_column", "password") > > > ># -- rr params -- > ># add value to ;lr param to make some broken UAs happy > >modparam("rr", "enable_full_lr", 1) > > > >modparam("auth_db", "db_url", "sql://serro:heslo@localhost/ser") > >modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser") > > > ># accounting > >#modparam("acc", "log_level", 1) > >#modparam("acc", "log_flag", 1 ) > > > > > ># ------------------------- request routing logic ------------------- > > > ># main routing logic > > > >route{ > > > > # initial sanity checks -- messages with > > # max_forwards==0, or excessively long requests > > if (!mf_process_maxfwd_header("10")) { > > sl_send_reply("483","Too Many Hops"); > > break; > > }; > > if ( msg:len > max_len ) { > > sl_send_reply("513", "Message too big"); > > break; > > }; > > > > # we record-route all messages -- to make sure that > > # subsequent messages will go through our proxy; that's > > # particularly good if upstream and downstream entities > > # use different transport protocol > > record_route(); > > # loose-route processing > > if (loose_route()) { > > t_relay(); > > break; > > }; > > > > > > ############# kvoli accountingu > > # labeled all transaction for accounting > > #setflag(1); > > # record-route INVITES to make sure BYEs will visit our server > too > > #if (method=="INVITE") record_route(); > > ##################################### > > > > # if the request is for other domain use UsrLoc > > # (in case, it does not work, use the following command > > # with proper names and addresses in it) > > ##if (uri==myself) { > > if (uri=~"zutom.sk") { > > > > if (method=="REGISTER") { > > > > # Uncomment this if you want to use digest authentication > > if (!www_authorize("zutom.sk", "subscriber")) { > > www_challenge("zutom.sk", "0"); > > break; > > }; > > > > save("location"); > > lookup("aliases"); > > break; > > }; > > > > # native SIP destinations are handled using our USRLOC DB > > if (!lookup("location")) { > > sl_send_reply("404", "Not Found"); > > break; > > }; > > }; > > # forward to current uri now; use stateful forwarding; that > > # works reliably even if we forward from TCP to UDP > > if (!t_relay()) { > > sl_reply_error(); > > }; > > > >} > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > -- > Jiri Kuthan http://iptel.org/~jiri/

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sr-users March 2004